[Asterisk-Users] Asterisk Passthrough

[Steve Creel]

On Tue, 2 Mar 2004, Emanuele Laface wrote:

>On Tue, 2 Mar 2004, Steve Creel wrote:
>
>> Your other option is to terminate the two PRIs into asterisk, and use
>> asterisk to provide two PRIs into your PBX.  This gives you access to the
>> actual call routing.
>
>Ok, my problem is exactly how I can do that?
>How can I say to asterisk "this port is connected to the world and the
>other is connected to my office telephon switch" (this is my main
>problem, I see something about groups but I'm not sure about the right
>configuration...)?

Don't try to map port to port - you're making your problem more complex
than it needs to be.  Let asterisk do some call routing for you.  You've
got an incoming call with dialed number identification.  Write an asterisk
extension rule to handle it... Should asterisk send it out on a specific
channel?  Should it be sent out to one channel out of a group?

For example, say your incoming number is 12345.  You've connected the
telco PRIs to spans 1 and 2, and your PRIs to the existing PBX are spans 3
and 4.  The telco channels are all in group 1, the channels to the PBX are
in group 2.

[incoming]
exten => 12345,1,Dial(ZAP/g2) ; Send incoming call for 12345 to the PBX


Now you have asterisk switching the call instead of cross connecting the
ports.

>How I can forward a call? It's simply an extension.conf rule?

Yes.

>When I make the forward in this way (with extension.conf rule) asterisk
>make some work or is a simple passthrough from interfaces?

Yes, it's some switching/callsetup work, but no codec translation, which
is by far your biggest CPU consumer.

>I need that calls "from PRI to PRI" don't load the computer.
>I want to use all CPU to (future) SIP calls.


Steve